Re: [asterisk-users] log incoming calls without answering

[Fabio Moretti]

Thank for all the replies, a lot of input and information!

Sorry for this useless mail, but I really wanted to say thank you.


Il 20/04/2017 17:26, Fabio Moretti ha scritto:
> Hi,
>
> I've some analogic lines and I'm asked if it's possible to program an 
> asterisk for "checking" the inbound calls without answering them, doing 
> something like this:
>
> analog line 1 -+-- asterisk
>|
>\__ analog phone
>
> when a call enter, asterisk sense it and store its values (callerid, date and 
> time, etc) somewhere, but nothing more, people will answer using the old 
> analog phone.
> The goal is to have a log of the inbound calls without touching the old 
> analog system (it's shared between different subjects).
>
> I'm pretty sure it's something possible, but how to tell asterisk: "ok, call 
> this AGI, and then don't answer and do nothing more".
>
> Any idea?
>
> Thanks
>
>
>
>   
>
>

-- 
Fabio Moretti
Gerente de Sistemas
www.tecytal.com <http://www.tecytal.com>
0800 8780
(+598) 248 77921


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Re: [asterisk-users] log incoming calls without answering

[Fabio Moretti]

Il 20/04/2017 18:09, kevin.lar...@pioneerballoon.com ha scritto:
>
> I honestly don't know if you can do what you want without some piece
> of equipment picking up the line. What I would do is get an analog
> line, an analog phone, an analog to sip device (there are many to
> choose from) and a basic asterisk instance. I would then make a small
> test setup where the analog line goes to a splitter. One side of the
> splitter goes to your analog phone. One side goes to your analog to
> SIP converter and then into your asterisk instance via your ethernet
> network. Use your cell phone to call the number of your analog line
> and see if it works. You would have to code a basic dialplan on the
> asterisk side and set up the trunk to your converter, which I am
> assuming you know how to do.

Yes, I'll definitely do the test before set up the whole proyect, but
the point basically is: it is possibile for asterisk to log a call
without answering it? How to do it in the dialplan? Or I'm wasting time
because an analog line who enter asterisk is always answered?







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Re: [asterisk-users] log incoming calls without answering

[Fabio Moretti]

Il 20/04/2017 17:32, kevin.lar...@pioneerballoon.com ha scritto:
>
> This gets kinda Rube Golberg-ish, but convert the incoming analog line
> to sip, route it through asterisk and have asterisk do its thing
> before converting it back to analog to send to the phone. Only problem
> is you get a lot of extra hardware involved in the mix to make it
> work. It will be a lot of expense and trouble, so you need to make
> sure that whatever part you want asterisk to play is worth that
> effort. Also, I wouldn't touch a fax line in this manner.
>
> If you could give a bit more info on what you want asterisk to do, we
> could maybe give better advice on how to solve your problem.

Hi Kevin,

I've already proposed your solution (is the most reasonable) but they
have more than 60 analogs lines (no faxes) and some of them terminate in
appliances like alarms, etc, so the solution must not touch in any way
the connection between the line and his termination: doing a analog to
digital conversion, passing it to asterisk and the convert it back to
analog is prone to problems (what if asterisk crashes? or if a gateway
fail?).
I can split the existing lines (there are no complex things like adsl or
digital signaling), convert the branches to digital and terminate then
into an asterisk machine, so any failure will not affect the old
circuit, but of course I've to configure asterisk to ONLY LOG calls and
nothing more.

This is what they want:
- line 1 ring
- line 1 is splitted in two, the first branch (let's say the "analog"
branch) go to an analog phone, that rings
- the second branch go through a gateway and then to asterisk
- asterisk log (with an AGI for example) "line 1 rings at  from "
no more is required from asterisk, if someone answer the analog phone or
not is not my business.







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[asterisk-users] log incoming calls without answering

[Fabio Moretti]

Hi,

I've some analogic lines and I'm asked if it's possible to program an asterisk 
for "checking" the inbound calls without answering them, doing something like 
this:

analog line 1 -+-- asterisk
   |
   \__ analog phone

when a call enter, asterisk sense it and store its values (callerid, date and 
time, etc) somewhere, but nothing more, people will answer using the old analog 
phone.
The goal is to have a log of the inbound calls without touching the old analog 
system (it's shared between different subjects).

I'm pretty sure it's something possible, but how to tell asterisk: "ok, call 
this AGI, and then don't answer and do nothing more".

Any idea?

Thanks






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[asterisk-users] strange warnings no samples for alawtolin

[Fabio Moretti]

[Aug 11 21:57:14] WARNING[1992] translate.c: no samples for alawtolin
[Aug 11 21:57:14] WARNING[2005] translate.c: no samples for alawtolin
[Aug 11 21:57:15] WARNING[2027] translate.c: no samples for alawtolin
Hi to all,

I have an elastix box running asterisk 1.8.20 without problem. It's
about four days I've started seen in log a warning message saying
translate.c: no samples for alawtolin, and now the frequency of this
message is about 6 times a second.

There's no other clue, everything is running smoothly and googling for
it doesn't help.

Here's an excerpt:

[Aug 11 21:57:15] WARNING[2029] translate.c: no samples for alawtolin
[Aug 11 21:57:15] WARNING[2038] translate.c: no samples for alawtolin
[Aug 11 21:57:15] WARNING[2045] translate.c: no samples for alawtolin
[Aug 11 21:57:15] WARNING[2055] translate.c: no samples for alawtolin
[Aug 11 21:57:15] WARNING[2059] translate.c: no samples for alawtolin
[Aug 11 21:57:15] WARNING[2078] translate.c: no samples for alawtolin
[Aug 11 21:57:19] WARNING[2093] translate.c: no samples for alawtolin
[Aug 11 21:57:19] WARNING[2095] translate.c: no samples for alawtolin
[Aug 11 21:57:19] WARNING[2110] translate.c: no samples for alawtolin
[Aug 11 21:57:19] WARNING[2120] translate.c: no samples for alawtolin
[Aug 11 21:57:19] WARNING[2125] translate.c: no samples for alawtolin
[Aug 11 21:57:19] WARNING[2132] translate.c: no samples for alawtolin
[Aug 11 21:57:19] WARNING[2139] translate.c: no samples for alawtolin
[Aug 11 21:57:19] WARNING[2141] translate.c: no samples for alawtolin
[Aug 11 21:57:20] WARNING[2152] translate.c: no samples for alawtolin
[Aug 11 21:57:20] WARNING[2174] translate.c: no samples for alawtolin
[Aug 11 21:57:20] WARNING[2177] translate.c: no samples for alawtolin
[Aug 11 21:57:20] WARNING[2208] translate.c: no samples for alawtolin
[Aug 11 21:57:20] WARNING[2210] translate.c: no samples for alawtolin
[Aug 11 21:57:20] WARNING[] translate.c: no samples for alawtolin

Does anyone have an idea of what is means and how I can get rid of it?

Thanks
-- 




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Re: [asterisk-users] CEL logging and queue APP_START/END, maybe an issue?

[Fabio Moretti]

  
  

Il 01/07/2013 15:17, Matthew Jordan
  scrisse:


  Nope, this is entirely expected.

  


  

  

[...snip...]

  

  On a side note, the fact that
masquerades are hard and tend to require people to do lots
of updates was a driving factor in the development efforts
that went on in 12. Masquerades are now an implementation
detail, so in the future, you won't have to deal with
BRIDGE_UPDATE.
  

  


ok matthew, thank you.
I understand, but now I'm getting a little confused: I think that
"linkedid" was the "long waited field" useful to follow a call
during its entire history (in fact I've made modification to my old
asterisk 1.4 dialplan to have something similar in the cdr using
accountcode field). Following what you say I should not only follow
the linkedid but, in case of a masquerading, I've to follow the peer
channel.
So, shoud I've to find and follow all the likedids related to every
BRIDGE_UPDATE? What if, for example, I've two ingoing call from
DAHDI that get bridged at some point? 
Is there a "correct" way to get all the records of a call in a way
that I can use to show the "history" of a call in a human readable
way? 
What I'm doing is experimenting, studying the CEL of an inbound
call-center, and due to the lack of documentation (and my lack and
experience) I can't understand how to follow correctly a call and,
for example, why rarely I get a BLINDTRANSFER, sometimes an
ATTENDEDTRANSFER and sometimes a FORWARD (I'm sure operators only
use the "TR" button on the phone). 
I think the most complete documentation is on wiki.asterisk.org, but
it's more like "XXX is when a channel is XXXed" than an explanation,
and there's no list of what apps can generate CEL events, when and
why.
I appreciate if you can point me to some document I've to study :)

Here I paste the events counts of my two-months CEL, maybe someone
can find it interesting:
ANSWER			166599
APP_END			42424
APP_START		42434
ATTENDEDTRANSFER	712
BLINDTRANSFER		15
BRIDGE_END		73575
BRIDGE_START		74325
BRIDGE_UPDATE		538
CHAN_END		1124624
CHAN_START		1124711
FORWARD			72
HANGUP			1124626
LINKEDID_END		54784

Thank you,
-- 
  

      
  
  Fabio
Moretti  
  Gerente de Sistemas 
   www.tecytal.com  
   0800 8780 
(+598) 248 77921  
  

  

  


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Re: [asterisk-users] Executing Script after MixMonitor is called

[Fabio Moretti]

.

  
  --Satish
Barot
  
  Ahmedabad,
India
  


  


  

  


  

  
  Some observations,
  
  (1) You are missing ^ in command
in Mixmonitor.In your case, It should be something like
MixMonitor(IND_PRI/${CDR(accountcode)}/OUT/${STRFTIME(${EPOCH},,%Y-%m)}/${STRFTIME(${EPOCH},,%d)}/${MIXMONITOR_FILENAME}.wav,m,/root/flac.sh
^${MIXMONITOR_FILENAME}.wav) 
  
  (2) You are passing just file
name as a parameter in your script and not a full path
for file. (Do you handle full path in a script?)


  --Satish Barot

Ahmedabad, India
  


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-- 
  

  
  
  Fabio
Moretti  
  Gerente de Sistemas 
   www.tecytal.com  
   0800 8780 
(+598) 248 77921  
  

  

  


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Re: [asterisk-users] Problem with CEL logging and channel bridging

[Fabio Moretti]

  
  

Il 13/06/2013 11:31, Fabio Moretti
  scrisse:


  Hi, I've already post this to the forum three days ago, sorry if it's
sounds like a crosspost, but I've got no replies, so I'm trying other
channels :)


ok, definitely CEL is a big question mark for most of us.

can someone point me to in deep CEL documentation or to an open
source code that use it so I can study more? not asterisk code,
please, I tried but I find really hard find how and then events are
generated.

thanks

-- 
  

  
  
  Fabio
Moretti  
  Gerente de Sistemas 
   www.tecytal.com  
   0800 8780 
(+598) 248 77921  
  

  

  


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Re: [asterisk-users] Problem with CEL logging and channel bridging

[Fabio Moretti]

  
  
Il 17/06/2013 11:13, Matthew Jordan scrisse:

  

  
Since you know that DAHDI/i1/96034296-30a3 is
  in a bridge with Local/1004@from-queue-00019c34;1 and
  Local/1004@from-queue-00019c34;2is in a bridge
  withIAX2/issuegroup-17175, you automatically know that
  DAHDI/i1/96034296-30a3 and IAX2/issuegroup-17175 can
  communicate (at least once everyone has Answered). The
  system you build on top of CEL has to understand the
  semantics of Local channels and tie the two together.
  
  
  
  Matt

  

matt, thank you very much. in fact I was wondering if
local-channel;1 and local-channel;2 have to be considered as "one"
channel or not.
Can I ask you if there's a in deep documentation of how channel and
events are generated/destroyed? I'm trying to find the time to
study, I'd like to generate a billing script based on CEL and a
graphical interface for visualizing calls history.

really thank you

-- 
  

  
  
  Fabio
        Moretti  
  Gerente de Sistemas 
   www.tecytal.com  
   0800 8780 
(+598) 248 77921  
  

  

  


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[asterisk-users] Problem with CEL logging and channel bridging

[Fabio Moretti]

Hi, I've already post this to the forum three days ago, sorry if it's
sounds like a crosspost, but I've got no replies, so I'm trying other
channels :)

This is the link to the forum post if someone prefer to reply here:
http://forums.asterisk.org/viewtopic.php?f=1t=86985

I'm using Asterisk 1.8.20.0 (the freepbx build) with CEL logging
activated. I'm using CEL because in our pbx we have different queues and
trunks serving different customers (we are an inbound call center) and
we need to detect when and how we have to bill our customers.
I'm facing an issue with the call transfer, for example I have:
- call entering a queue
- operator answer the call
- operator make an outgoing call to reach the customer
- operator put in communication the ingoing call with the outgoing
this result in various channel to be created/destroyed, and I'm using
bridge events to detect what is going on with the call. In this case I
have (I've hidden CHAN_START,ANSWER and HANGUP events because they have
no useful information in this case):

++---+-+---+-+--+-+-+--+

| id | eventtype | eventtime   | exten | context | 
channame | appname | appdata | peer 
|

++---+-+---+-+--+-+-+--+

| 965224 | BRIDGE_START  | 2013-06-10 10:15:18 | 20| ext-queues  | 
DAHDI/i1/96034296-30a3   | Queue   | 20,t,,  | 
Local/1004@from-queue-00019c34;1 |

| 965226 | BRIDGE_START  | 2013-06-10 10:15:18 | s | macro-dial-one  | 
Local/1004@from-queue-00019c34;2 | Dial| SIP/1004,,trM(auto-blkvm) | 
SIP/1004-40ce|

| 965340 | BRIDGE_UPDATE | 2013-06-10 10:16:08 | s | macro-dialout-trunk | 
Local/1004@from-queue-00019c34;2 | Dial| IAX2/issuegroup/110,300,| 
IAX2/issuegroup-17175|

| 965513 | BRIDGE_END| 2013-06-10 10:18:15 | 20| ext-queues  | 
DAHDI/i1/96034296-30a3   | Queue   | 20,t,,  | 
Local/1004@from-queue-00019c34;1 |

| 965515 | BRIDGE_END| 2013-06-10 10:18:15 | s | macro-dialout-trunk | 
Local/1004@from-queue-00019c34;2 | Dial| IAX2/issuegroup/110,300,| 
IAX2/issuegroup-17175|

++---+-+---+-+--+-+-+--+


The first BRIDGE_START is the connection between the inbound call
(DAHDI/i1/96034296-30a3) and the local phone
(Local/1004@from-queue-00019c34;1), the second BRIDGE_START is the
connection between the local phone (Local/1004@from-queue-00019c34;2)
and the outgoing call (SIP/1004-40ce) that is going out by a IAX trunk.
After that I have a BRIGDE_UPDATE event where no field make me know
which channel is being updated, I only have the channame
(Local/1004@from-queue-00019c34;2) that is the channel being bridged out
and the outgoing channel (IAX2/issuegroup-17175), but I have no
information that in fact the ingoing call (DAHDI/i1/96034296-30a3) is
being bridged to the outgoing channel.
I have no other event (TRANSFER or something like that) to know what is
going on.

In my cel.conf I have:

apps=queue
events=CHAN_START,CHAN_END, APP_START,APP_END, ANSWER,HANGUP,
BRIDGE_START,BRIDGE_END,BRIDGE_UPDATE,
BLINDTRANSFER,ATTENDEDTRANSFER,TRANSFER, PICKUP, FORWARD,
PARK_START,PARK_END, LINKEDID_END

Should I change something in my configuration or it's wrong to rely on
bridges to follow a call? What kind of event should I follow to be sure
to catch where the call is going?

Thank you for any suggestion!


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Re: [asterisk-users] A problem with IAX2

[Fabio Moretti]

  
  
hi,

I've solved various iax2 problem mentioning calltoken when I put
these lines in the iax configuration:

requirecalltoken=no
calltokenoptional=0.0.0.0/0.0.0.0

bye

Il 11/06/2013 19:25, Mordechay Kaganer
  scrisse:


  B.H.
  On Jun 11, 2013 5:15 PM, "Steve Totaro" stot...@totarotechnologies.com
wrote:




 On Tue, Jun 11, 2013 at 8:32 AM, Mordechay Kaganer mkaga...@gmail.com
wrote:

 B.H.

 Hello!

 We have several Asterik boxes that are connected to
PSTN using PRI cards and they are interconnected using IAX2
trunks so that incoming calls are delivered from PSTN to the
servers they belong to.

 In past we were using asterisk 1.4 on the server that
is receiving IAX connections and everything worked as expected.
Recently, we have switched to a newer box with asterisk 1.8.22
and then we began to experience sometimes a strange problem:

 At some point of time, incoming IAX connections begin
to get refused by the server and we get the following messages
in the logs:

 WARNING[] chan_iax2.c: Too much delay in IAX2
calltoken timestamp from address X.X.X.X

 where X.X.X.X is the IP of the PSTN-IAX gateways
and all the incoming calls start to be rejected.

 Direct PSTN calls (both incoming and outgoing) to the
same server work OK. The only solution that helps is to kill the
asterisk and restart it.

 All the servers are connected to the same LAN segment,
with gigabit switch, there is no problems with the network. No
packet loss.

 There's already bug report present with very similar
issue, but it is "suspended" and, like stated there, the problem
is very hard to reproduce.

 See:https://issues.asterisk.org/jira/browse/ASTERISK-21762


 -- 
  NOW!


 Use SIP and never look back.

 Thanks,
 Steve Totaro

 --

  Thanks, that's what i actually going to do.
  But does this mean that IAX is obsolete? Actually i have
selected IAX in the first place because it looks like more
"native" for asterisk, so i thought it would be more suitable as
a protocol to interconnect asterisk boxes...
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-- 
  

  
  
      Fabio
Moretti  
  Gerente de Sistemas 
   www.tecytal.com  
   0800 8780 
(+598) 248 77921  
  

  

  


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